Re: [Asterisk-Users] IAX UDP packet dropped on incoming call
It's probably because the hole in your firewall has closed. Either increase the amount of outbound traffic through that port (thus keeping the association alive), or modify your firewall to have a fixed port mapping to your asterisk box. -brian Gene Willingham wrote: When receiving incoming calls, I periodically get a UDP packet dropped message on my firewall. This prevents the incoming call from completing. It appears to be a random occurrence, sometimes hours, sometimes half hour, sometimes minutes. I am using Asterisk 1.0.1 if this helps. Gene ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 SIP 7.2 keypress (not DTMF) problem
This used to happen in 6.3 all the time for me. I upgraded to 7.2 hoping that it was one of the things they fixed. But alas it wasn't. It's interesting that the key events are getting recognized enough to produce the tone feedback, but that those events are not being properly communicated to other parts of the software. Makes me really curious about the SW architecture of this thing. -brian Marty Mastera wrote: Since upgrading to 7.2, I've noticed a random problem where I dial a number and hear all the correct tones in the handset, but the display won't show all the numbers I dialed. So you sit there waiting for the dialplan to kick the call off (b/c you heard the proper amount of tones played and think it's all good) but the phone is just sitting there b/c it somehow missed digits. (For example, I dial 93035551212 and hear the correct DTMF in the handset, but the display shows 9303551212) It doesn't seem to be digit specific, and can lose one or more digits when the problem happens. Dialing very slow and deliberate seems to help, although I haven't done super serious testing of that yet... Any ideas? Marty Mastera M3 Resources [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Phone: 303.680.1283 x200 FAX:303.680.1283 IAXTel: 700.206.7507 FWD: 484162 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RC1 still broken with Cisco 7960?
After downloading the latest CVS head and testing it with the Cisco 7960 (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid audio dropouts. I'm quite sure my gateway provider is running an older version of Asterisk, and I suppose that this may be the root cause. But I mention the issue here because it seems like it would be a mistake to ship Asterisk 1.0 if it doesn't work properly with Cisco phones (as there are undoubtedly a lot of them out there). -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio level in compressed wav files
Unfortunately, this doesn't really work out to be a great solution. The dynamic range of the original recording is limited and scaling it after the fact just yields fairly distorted sounding recordings. It seems like the problem is a bug in the implementation of the compressed file format encoders, or the process by which they get invoked. I guess I'm going to have to dig into the code. -brian Bill Seddon wrote: Brian Take a look at sox (type man sox at the command prompt if sox it installed for details on the options available). There is a vol argument that allows you to adjust the gain. If this is what you need, you can call sox after record (using the system command) to adjust the gain. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: September 11, 2004 7:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Audio level in compressed wav files Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the wav format are about twice the level of those made using WAV or wav49. Unfortunately, the wav recordings are uncompressed and about 10 times the size of the other formats. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio level in compressed wav files
Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the wav format are about twice the level of those made using WAV or wav49. Unfortunately, the wav recordings are uncompressed and about 10 times the size of the other formats. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 dropping loop current 10 seconds after answer
Hi everyone, I have a TDM400 configured with 4 FXS ports, each connected to a caller-id analog trunk port on a Nortel system. Outgoing calls work great. But on incoming calls it appears that loop current is getting dropped momentarily about 10 seconds after the call is answered. Since the Nortel system is programmed to recognize this as remote party hangup it is causing all incoming calls to get dropped almost immediately. Changing from ks to ls in * doesn't make the problem go away. Any thoughts? Thanks -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP hackers gut Caller ID
The real problem here is that people shouldn't be using callerid as an authentication scheme. Lots of people have had the ability to set arbitrary clid for years and yet banks and other institutions have stupidly used it to authenticate callers. Complaints should be directed to them and not the VoIP industry. -brian Alex wrote: Here is what you can possibly do: - Steal calling cards if they are useing caller id authentication scheme - Get access to personal banking information (Citibank uses callerid as part of authentication process.) - Purchase goods and services backed up by calling verification. I can go on and on for hours. Main point of story that [EMAIL PROTECTED] will hit the fan and VOIP will be regulated badly. Especially if some known terrorist will confess about using Vonage in Afaganistan.or some of drug dealers/weapon traders will be cought . Bug generraly author of that article is an idiot. He does not understand the difference beteween VOIP and ISDN PRI. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listas iPfone Sent: Wednesday, July 07, 2004 6:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID This is very interesting... Regulations..USA... But... what can i do faking a caller id? stolen what? what is the point? miklos - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 12:56 PM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind Iptables: What's the magic?
Which way is the audio working? -brian Isamar Maia wrote: I tried some combinations of setup seen in some postings and didn't get success on this yet. I have grandstream phones outside the network trying to call an * server inside my network through NAT/Iptables. The problem that I'm facing is one-way audio. Any suggestion? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI with National (north america) Signalling
Actually, here in Maryland ISDN BRI is cheaper than POTS. POTS business lines are like $20 each, and caller-id is around $8.50 per line. So two lines with caller-id are about $57. On the other hand, a BRI, which has awesome voice quality and includes CLID is $45. If you get a residential ISDN line they can be as cheap as ~$30. And I agree with whomever said that Verizon doesn't quite get it. Whenever I call about an ISDN line they try really hard to steer me towards DSL. Although if you get to the right business unit they're a little better. Years ago I had a bunch of ATT 7506 phones on a BRI with CO-based custom ISDN centrex. It was like having my own $20M switch. Of course convincing them that they *could* do this, and that it was a tariffed service was difficult. There were times I had to fax them copies of the relevant ISDN tariff and pages from the 5ESS provisioning guide. ISDN can be very, very cool. The Europeans have figured this out, but the US telcos are just waiting to be put out of their misery. Hopefully VoIP will do it. -brian Walt Reed wrote: On Wed, Jun 09, 2004 at 11:24:11AM -0400, Jon Pounder said: Also for any ISDN gurus out there - is there a simple way to loop back BRI so I can call from one B to the other for testing with the proper signalling for National to see if asterisk actually works without committing to ordering a line that will be useless if it does not work. While I'm not an ISDN guru, a google for ISDN loopback shows products in the $150 range that are designed for this. Most seem to be euro, but there are US products too. FWIW, I would also be very interested in US BRI ISDN w/ * info. Analog POTS just blows. Looking through the Verizon tariffs, it seems as conversion from POTS to BRI is supported and reasonably affordable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI In the states
Scott Nelson wrote: On Monday June 7 2004 09:22, Daniel Jimenez wrote: No one has any comments on this? No recommendations, or you are stupid for trying that or anything? How about, I'm interested too? I've had an ISDN line for ages, but I've always used a Terminal adapter. Now that my office is looking into VOIP options and I'm the guinnea (spelling?) pig, it would be nice to combine everything with Asterisk. I think. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've been running Nortel switches here in the states with BRI trunking for more than a decade. I am constantly told that the audio coming out of our office is nothing short of stunning. BRI is a great thing for voice, although in this country the phone companies just can't stop thinking data when you talk to them about it. Would really like to stick an Asterisk box between the CO and the Nortel switch if anyone comes up with a BRI card that works in the US. Another problem is that all the Euro-BRI cards are ST interface. Here in the US BRI is delivered from the LEC as a U. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hot keypad on a Cisco 7960
Matt Darnell wrote: Aloha, Does anyone know how to have a hot keypad on a Cisco 7960? It allows you to dial on-hook without press the SPEAKER button. Very handy once you get used to it! I've been looking for the same thing. My Nortel system does that, and it is very addictive! -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura-SPA2000 background noise
Nicolas Gudino wrote: Hi Brian, Brian Cuthie wrote: BTW, anyone know how to get the SPA-2000 do drop loop current momentarily when the other end hangs up? -brian There is a web configuration option to reverse the polarity in the latest 2.0 firmware. Yeah, I saw that too. But it doesn't always seem to fire when I think it should. And, my Nortel switch ignores it anyway, since they have conveniently made their trunks polarity insensitive. What would be better is if it dropped loop current entirely for a few hundred milliseconds. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Transfer with Budgetone
Adam Goryachev wrote: On Thu, 2004-06-03 at 11:40, Tony Hoyle wrote: Adam Goryachev wrote: Well, actually they are. Sure, for $20 you can buy an analog phone, for $150 you can buy a grandstream, big difference. However, for a PBX class telephone, you are looking at prices $500 per handset No idea what you mean by PBX class telephone but if anyone at our company spent $500 on a phone they'd probably be fired (unless it was the boss). ie, new phones from NEC or other proprietary phone systems... Where in the world are you buying your phones? New phones for any PBX (Nortel, Lucent, Toshiba, etc.) range from $100 to $350 for all but the most esoteric models. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura-SPA2000 background noise
Shaun Ewing wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Tuesday, 1 June 2004 10:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura-SPA2000 background noise I have been using Cisco ATA's for analog connections and decided to give a Sipura SPA-2000 a try. I noticed there is a fair amount of background white noise that is noticeable, especially after breaking the dial tone. After pressing a '1' to break the dial tone, there is a fair amount of noise that is evident. I do not notice this condition on the Cisco ATA's. I plugged the Sipura in the same location as the Cisco ATA. Anyone else have this condition with the Sipura? Mine started doing this, and over the next few days it got so bad that the device was unusable. Sipura didn't respond to the emails from me or the reseller whom I purchased the device from, but the reseller offered a refund. I'm not sure what ended up happening from the reseller's end once I returned it. -Shaun I have a fairly new Sipura 2000 that also is very noisy. Quite disappointing, actually. BTW, anyone know how to get the SPA-2000 do drop loop current momentarily when the other end hangs up? -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unblocking incoming SIP
Eric Wieling wrote: On Mon, 2004-05-31 at 10:16, Duane wrote: Andy Powell wrote: Anything that's added to * that breaks how protocols work should be by default OFF not ON, but that's just IMO... I agree 100%, this has been very frustrating trying to work out why Asterisk suddenly stopped accepting calls from FWD and other PSTN based services. I very much preferred the old method, if I didn't want to accept a SIP call you just don't have a matching context. The problem is that too many people had a context= in [general] and didn't realize that incoming SIP calls that didn't match anything would be accepted and sent to the context= that was specified in [genera]. Which is because configs/sip.conf.sample has context=default. So let's not blame it on the too many people problem. I agree that new features shouldn't break old configs. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unblocking incoming SIP
Andy Powell wrote: On 31/05/2004 at 10:47 Eric Wieling wrote: On Mon, 2004-05-31 at 10:16, Duane wrote: Andy Powell wrote: Anything that's added to * that breaks how protocols work should be by default OFF not ON, but that's just IMO... I agree 100%, this has been very frustrating trying to work out why Asterisk suddenly stopped accepting calls from FWD and other PSTN based services. I very much preferred the old method, if I didn't want to accept a SIP call you just don't have a matching context. The problem is that too many people had a context= in [general] and didn't realize that incoming SIP calls that didn't match anything would be accepted and sent to the context= that was specified in [genera]. which is why everywhere you look in the guides etc people say put something like: context=boguscalls in the general section, which (providing you weren't stupid enough to create a [boguscalls] section worked well... in fact I'll go as far as quoting my own guide: An important point here, if you do not have a sip aware firewall and are just using port forwarding then ensure that your context points to somewhere like invalidcalls. If you do not do this then someone could call one of your extensions direct from the Internet. If you had an FXO card in the machine, this could lead to them being able to make PSTN calls Those people that didn't realize were more than likely using a guide to set up... I still stand by the fact that this feature should have been OFF in the first place. Andy Except that I *want* anyone to be able to call me directly from the Internet. That's the whole point -- we're trying to remove the necessity for a phone-company-like entity in the middle. Instead, I suggest setting the default context for sip to something like sip-incoming-default and then include in the dialplan those things you wish people to be able to call directly. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
John Todd wrote: At 7:33 AM -0700 on 5/26/04, Maveric wrote: I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? [catching up on 800 -user posts - sorry for delay] Nobody on the list suggested this method that I saw: Use the Background application, but play silence. You'll notice in the asterisk-sounds directory (the additional package) there is a directory called silence which contains 10 files ranging from 1 to 10 seconds of silence. Works the same as Wait from the user's perspective (they hear nothing) but lets the user type keys. That's why I made those files; it's only a slightly ugly hack, and it works quite well. :-) Only slighly ? :-) -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PostgreSQL
Fabio, You need to enable tcp connectivity on psql. Wherever you configured the databases to live (/var/lib/pgsql/data on my machine) you'll find a file called postgres.conf. You need to read that and uncomment out the appropriate lines to get: tcpip_socket = true port = 5432 -brian Fabio Donaggio wrote: Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura
Bruce, I think this is related to your firewall. You may want to take a look a posting I did a few weeks ago. http://lists.digium.com/pipermail/asterisk-users/2004-May/046511.html Something on this topic probably belongs in the wiki. -brian Bruce Komito wrote: I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call terminates immediately because I am watching the CDRs come out. The * server is on a public address with no firewall between it and the outside world. sip.conf: (both extensions have identical settings) ; Bruce [5815] type=friend username=5815 secret=wpti5815 host=dynamic [EMAIL PROTECTED] context=vpbx-wpti qualify=3000 dtmfmode=inband disallow=all allow=ulaw allow=alaw nat=yes I'm thinking this has something to do with a setting in the Sipura, but I don't know where to start. I have nat keep-alive turned on, but I had to turn stun off because it was causing a long, inexplicable delay after dialing before the call would complete. I'm realizing NAT with VoIP is a real problem. Anyone have a silver bullet they wish to share? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura
Please ignore my previous post (below), as it's not really relevant to your problem. I was in some kind of mindless auto-email processing mode and responded without fully reading your message. Too much spam, too little sleep. Geesh. -brian Brian Cuthie wrote: Bruce, I think this is related to your firewall. You may want to take a look a posting I did a few weeks ago. http://lists.digium.com/pipermail/asterisk-users/2004-May/046511.html Something on this topic probably belongs in the wiki. -brian Bruce Komito wrote: I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call terminates immediately because I am watching the CDRs come out. The * server is on a public address with no firewall between it and the outside world. sip.conf: (both extensions have identical settings) ; Bruce [5815] type=friend username=5815 secret=wpti5815 host=dynamic [EMAIL PROTECTED] context=vpbx-wpti qualify=3000 dtmfmode=inband disallow=all allow=ulaw allow=alaw nat=yes I'm thinking this has something to do with a setting in the Sipura, but I don't know where to start. I have nat keep-alive turned on, but I had to turn stun off because it was causing a long, inexplicable delay after dialing before the call would complete. I'm realizing NAT with VoIP is a real problem. Anyone have a silver bullet they wish to share? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse SIP
I'm using Coloco now, which so far is working well. Where companies like VoicePulse buy services from a patchwork of CLECs in order to cover their markets, Coloco is a CLEC. The upside is that you cut out the middleman. But if you need a number in an area they don't serve you'll need to find a different provider. Coloco serves latas 236 and 238 (NPAs 301,240,410,443,703), which works well for me since I'm in 238. If you need numbers local to DC and central Maryland give them a shout (coloco.com). I hear they're also working with some other CLECs to get numbers in other areas but I don't have any details on that. -brian David H Hickman wrote: Who do you use now? David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 On May 21, 2004, at 8:49 PM, Brian Cuthie wrote: SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic SIP.CONF
So I've been kind of struggling with the notion of making my Asterisk
implementations dynamic, too. While I'd like to make everything directly
database driven, I'm not sure Asterisk is quite there yet.
I've been thinking of writing something that creates appropriate
configuration files from the database on a periodic basis, and then does
an Asterisk reload. This would introduce a small delay into
configuration changes, but it does have other benefits such as
decoupling the design of the database from Asterisk.
Any thoughts?
-brian
Fran Boon wrote:
Darren Nay wrote:
We are looking to expand our usage of Asterisk and I am trying to
make as
much of the configuration dynamic as I possibly can. The only part
that I'm
having problems with is sip.conf. I can get asterisk to register each
extension with our local SER SIP proxy dynamically by using the
sipfriends table in the database, but I'm having trouble with the
message
waiting indicators (ie. SIP NOTIFY packets when a new voicemail is
waiting).
-SNIP-
Is there a way to make this dynamic so that I don't have to add this
into
sip.conf -every- single time that I add a new extension?
Only by extending the functionality of sip friends to include this
extra field...
I wouldn't bother doing this as ast_data (formally res_data) is being
developed to replace sip/iax friends.
If you want to take a sneak preview at this then see:
http://svn.asteriskdocs.org/res_data/ast_data/
I tried the following, but it didn't work ..
[default]
type=peer
host=dynamic
dtmfmode=inband
username=${EXTEN}
Mailbox=${EXTEN}
Am I on the right track, or way off base? :-)
Way off base ;)
That kind of syntax only works in extensions.conf
F
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Re: [Asterisk-Users] fwd on busy when calling multiple extensions at once
You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk upgrade on production box
I'm sure there are better ways, but I usually do the following: 1) make sure my current source directory is named something else (see step 3) 2) fetch cvs head 3) mv asterisk to something like asterisk_cvs_head_5_21. This keeps all the old source trees around so that I can easily roll back to any version I've installed previously. 4) cd to the new asterisk directory (whatever you called it in step 3) 5) make 7) asterisk -r 8) show channels to make sure nobody's using it 9) stop now 10) exit asterisk 11) make install 12) restart asterisk Note that if you don't 'make samples' the stuff in /etc/asterisk won't get torched. This should be fine, assuming that the new version doesn't require any config changes. Naturally, you'll want to poke around in asterisk/configs to see what kind of new options are available. -brian Nik Martin wrote: What is the best way to upgrade a production asterisk box? make upgrade? I don't want my configs messed with, and need the process to go as smooth as possible. I fetched and built a new kernel last night, but haven't rebooted into it. I'll do that tonight, and then want to quickly upgrade to the latest asterisk (mainly for zttest.) Does make upgrade fetch head? Thanks Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP timestamps
Excellent! I'll give it a shot. -brian brian k. west wrote: This time stamp issue is all gone.. now if everyone will just UPDATE! bkw - Original Message - From: Andres [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 21, 2004 4:14 PM Subject: Re: [Asterisk-Users] RTP timestamps VC: My phone is broken: I get no audio. TAC: Show us a network trace. VC: (presents my ethereal traces, with the non-counting RTP timestamps) TAC: (laughing) NEXT!!! I have not read RFC1889 (RTP) in detail, but I am positive that the timestamp field was put there for a reason. Sure, maybe Cisco is a little overzealous in the way their code handles non-conformance, but to try and put the blame entirely on them is misdirection. My ATA-186 has problems with the same RTP stream. GIGO. * needs to generate RTP streams with valid timestamp progression -- surely we're not happy to say the Cisco 79x0 is the only phone that cares about timestamps, so there's the problem. Hi Vic, For your information Sipura also suffers from the Timestamp issue. 3 months ago when I opened the case with them, they explained in detail why they needed those Timestamps (it has to do with the jitter buffer calculation algorithms). They told me the problem had to be solved at the Asterisk side since there is no reason why the Timestamps should change. They have not seen this weird behaviour with any other SIP system besides Asterisk. In any case, thats why we came up with the rtp.c hack, and have been happy ever since. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse SIP
SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote: Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a Service Unavailable back from gw5.voicepulse.com. If I try IAX2 with the same settings, the call goes through - but sound is horrible. Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
brian k. west wrote: You know, I'm not so sure this is limited to chan_capi. I have two asterisk boxes running, with one connected to my PSTN gateway (also using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head works if I comment out the offending lines. Without commenting them out, the cisco phones drop packets like crazy. No chan_api is involved. I use two 7960s daily with cvs-head out two cisco gateways and in and out via nufone IAX2 without issues. The question is what is diffrent about your setup vs mine? That's a good question. Let me describe my setup and when the problem occurs and when it doesn't and maybe it'll help in figuring out what's really going on here. I have something that looks like this: [PSTN Gateway *] -- 100Mb Ethernet -- [Systemix * svr] -- public Internet -- [local * server] -- 100Mb Ethernet -- [Cisco 7960] (All inter-Asterisk communication is done using IAX) In this configuration with the cvs head from 5/9 the Cisco phone works horribly (dropped/repeated? packets) unless I comment out the timestamp check in rtp.c. Stable seems to work pretty well. Now if I change the config so that it looks like this: [PSTN Gateway *] -- 100Mb Ethernet -- [Systemix * svr] -- public Internet -- [Cisco 7960] Things seem to work ok without having to make changes to rtp.c. So I've had a suspicion that it has something to do with delay or jitter that's introduced in the IAX channel that goes over the public Internet. But, as you say, this works fine for you. Could it be the extra * in the middle? That's one difference between our setups. Now the big unknown here is which version of * my PSTN provider is running. I've asked them. I'm hopeful I'll eventually get an answer. And as far as 'fixing' it goes, I would love to. I'm not without the skill. But, while Asterisk is almost unbelievable in its features set, some of the code is damn hard to grok. Some source files have as many as 8000 lines with virtually *no* comments. I don't think I've seen a single function with a preamble describing what it does, or how it works. Look at the .h files in include/asterisk/ I have, and while there is some documentation there, it's limited and does little to explain the actual code. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
brian wrote: You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. No I'm getting it loud and clear. You have some IAX providers that do not want to take care of customers when the software they use to provide service to their customers needs an update they refuse or fail to upgrade. Not our problem if they choose not to. If they update to cvs-head the problem will go away and its backwards compatible with cvs-stable. You can continue to hack rtp.c or ask your providers to upgrade. If they refuse to take care of you then I would consider getting service elsewhere. Yeah, right. And just how often should my service provider update their Asterisk installation? Daily? Weekly? Frankly, given that there's not even a 1.0 release of Asterisk I'm amazed any service providers are using it. Asterisk is currently a rapidly moving target, as this very issue demonstrates. Once a 1.0 or 1.1 is released you can bet everyone will upgrade. Until then, all of us should probably keep our expectations in check. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Kevin Walsh wrote: Brian Cuthie [EMAIL PROTECTED] wrote: Also on a side note if Kapejod isn't wanting keep chan_capi up to date then someone needs to ask him if he will disclaim it so digium can include it and help maintain it. I'm almost certain I didn't say this. Please be careful with your attributions. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940
Graham,
You need to configure something in extensions.conf to access voicemail.
I usually use something like this:
exten = 8500,1,VoiceMailMain(s${CALLERIDNUM})
exten = 8500,2,Congestion
Then you'll want to configure the voicemail URI on the 7940 so that it
calls extension 8500.
One nice thing about the Cisco phone is that they will keep track of WMI
separately for each configured line.
-brian
Graham Turner wrote:
can anyone give me a reference to the retrieval of voicemail from the
Asterisk PBX using a cisco 7940 phine running sip image.
i have configured a single voicemail box using the script, the corresponding
entry in voicemail.conf and configured the extension to use the voicemail
box .
i can see from the asterisk console the message being passed to the voice
mailbox, and correspondingly the sip phone indicates voicemail by the
flashing red on the handset and the envelope on its console
it would seem further configuration work is required to access the voice
mailbox
TIA
GT
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Re: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940
brian wrote: i have got exten = 1001,2,Voicemail(u1001) This is for leaving voicemail. VoiceMailMain is for you to check voicemail. i know there has been recent developements to the voicemail application but is this correct given a cvs download of early this month ?? It hasn't changed how you check/user voicemail. 2nd qu - where do i configure the 'voicemail uri' - have been through the phone / line settings - or do i have to configure the SIPMAC or sipdefault.cnf files ?? I think you can do it via the phone.. I have always done it in the .cnf files. It's in the SIP Configuration part of the phone setup. (Of course this assumes you're using the SIP image.) -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Iain,
This is a known issue with the Cisco phone and Asterisk having to do
with a change made later in the cvs tree. Try 1.0 stable, or modify
rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */
rtp-lastts = rtp-lastts + ms * 8;
// if (!f-delivery.tv_sec !f-delivery.tv_usec) {
/* If this isn't an absolute delivery time,
Check if it is close to our prediction,
and if so, go with our prediction */
if (abs(rtp-lastts - pred) 640)
rtp-lastts = pred;
else {
ast_log(LOG_DEBUG, Difference is %d, ms
is %d\n, abs(rtp-lastts - pred), ms);
mark = 1;
}
// }
} else {
This seems to work for me. Others may have more insight.
-brian
Nik Martin wrote:
Out of context, this isn't much information. Is your network connection OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Tuesday, May 18, 2004 1:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling:
Cisco 7960 = asterisk = IAX
produces sound drop outs so extreme that the call is useless.
I noted this
in an earlier post. Dialling:
Cisco ATA186 = asterisk = IAX
is fine.
Frankly, I think this is such a bad problem that it should be
sorted in
advance of any of the new features that seem to be getting
such prominence
nowadays. It was not present earlier in the year and I
haven't upgraded my
7960. So I don't think you can point the finger entirely in Cisco's
direction.
Iain
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Re: [Asterisk-Users] AArgh, * and the 7960
You know, I'm not so sure this is limited to chan_capi. I have two asterisk boxes running, with one connected to my PSTN gateway (also using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head works if I comment out the offending lines. Without commenting them out, the cisco phones drop packets like crazy. No chan_api is involved. And as far as 'fixing' it goes, I would love to. I'm not without the skill. But, while Asterisk is almost unbelievable in its features set, some of the code is damn hard to grok. Some source files have as many as 8000 lines with virtually *no* comments. I don't think I've seen a single function with a preamble describing what it does, or how it works. And I don't mean any offense by this. As I said, Asterisk is a truly amazing piece of software. But if the original developers, who really know how this stuff works, could put some effort into documenting the code with some comments, their efforts will pay off ten-fold when others are able to start helping them maintain it. Cheers, -brian brian k. west wrote: Also on a side note if Kapejod isn't wanting keep chan_capi up to date then someone needs to ask him if he will disclaim it so digium can include it and help maintain it. bkw - Original Message - From: brian k. west [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 18, 2004 8:01 PM Subject: Re: [Asterisk-Users] AArgh, * and the 7960 I'd love to fix the problem, but no-one is listening! I did what you said, captured Ethereal traces, found that timestamps do not increment, found BLATANT errors in rtp.c where a signed int is being used to hold return values from an unsigned int function... and had my bug report thrown out because I am only able to reproduce the problem with chan_capi. The problem isn't with asterisk chan_capi will have to be updated to deal with the changes. Now I know that chan_capi doesn't belong to Digium, and I know that you're all trying to get a 1.0 release out. But this problem is really hurting my business, and right now destroying any chance that I might start offering Asterisk as part of commercial solutions. I don't see these issues in any other channel driver. Now, kapejod is not replying to my e-mails, and markster's suggestion (from another bug report) of zeroing out the delivery field in chan_capi's read function did not work. So hacking is all I have left if I want to keep using Asterisk -- which I do, because I think it's a great program with a pretty good community around it. Where are you ethereal traces so I can look over them. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftPhone to SoftPhone with No Voice
Do you have iptables turned on with rules that restrict packets to the RTP ports? Try doing an iptables --flush then see if it works. If so, you'll need to open up the UDP ports that RTP is configured to use. Normally Asterisk would open up the firewall by sending packets out those ports, but when both sides of the call live outside the firewall (iptables in this case) and they're both SIP you'll have this problem. See my previous posting for a more detailed explanation (or email me directly). -brian [EMAIL PROTECTED] wrote: Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf and extensions.conf. Let me know if i missed something. Thanks Deepak sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = from-sip ; Default context for incoming calls ;srvlookup = yes; Enable DNS SRV lookups on outbound calls ; Asterisk only uses the first host in SRV records ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility ;tos=lowdelay ; IP QoS parameter, either keyword or value ; like tos=184 ;maxexpirey=3600; Max length of incoming registration we allow realm=asterisk ; Our global authentication realm ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs allow=all ; Allow codecs in order of preference ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc . [Phone1] type=friend host=dynamic defaultip=192.168.3.103 dtmfmode=rfc2833 context=from-sip callerid= Win box 1 [Phone2] type=friend host=dynamic defaultip=192.168.3.119 dtmfmode=rfc2833 context=from-sip callerid= Deepak 2 [Phone3] type=friend host=dynamic defaultip=192.168.3.106 dtmfmode=rfc2833 context=from-sip callerid= Ravi 3 [extensions.conf] [from-sip] exten=1,1,Dial(SIP/Phone1,20,tr) exten=2,1,Dial(SIP/Phone2,20,tr) exten=3,1,Dial(SIP/Phone3,20,tr) This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loop length supported by FXS module on Wildcat TDM400 card
Steven Critchfield wrote: On Fri, 2004-05-14 at 16:12, Elliot Eichen wrote: Does anyone know (appoximately) the max loop length that the FXS module on a Wildcat TDM400 card will drive an analog phone(over standard - say cat3 or 26 gage twisted-pair wire). It's clearly not going to be 18kft, but perhaps 4000 feet? I'd be wary of 4kft even if it could do it. Thats a long way for something to short/introduce power and fry your complete computer. It would be pretty easy to put a cheap pc local and run a few feet of cable. This is especially true if you run more than one line. Shorts won't be a problem since any reasonably designed FXS interface has current limiting. In fact, it's this current limiting and the relatively low loop voltage (I'm guessing 24V on a card of this type) that will restrict usable loop length. As you point out, induced transients are also a problem. So if you have a long loop make sure you put telco surge arrestors on each end. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 - licenses and opinions
BZZZT! Wrong too. Patents are a trade. The holder of the IP opens it up for public scrutiny and in return for exclusive control. Otherwise, companies would (and often do) keep the IP a trade secret. -brian Andrew Kohlsmith wrote: Just remember that you were given those patents as incentive to invent so that ultimately your work would go into the public domain so we can all enjoy it. We are buying your work with our tax dollars by protecting it for a short period of time so you have a little monetary incentive. BZZZT! Wrong. He was given those patents as in incentive to invent something that he could SELL without everyone on the planet copying his hard work and competing on his idea. Patents put the process out in the public so that it's easy to see when someone's infringing. 17 years for software patents is FAR too long, IMO, but that's an entirely different story. IMO software patents shoudln't be for more than ~24 months since the industry moves so blazingly fast. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
Alex, The media ports are configured in rtp.conf. Also, note that Asterisk sends RTP packets out the same ports it expects them to return on. This has the effect of creating a NAT mapping for that 5-tuple, as well as opening a hole in your firewall (naturally, YMMV depending on exactly what you're running for a firewall). One interesting consequence of the way Asterisk works is that if you don't have anything behind the NAT/Firewall that's generating RTP packets (ie, no audio) no hole gets made and incoming packets will get rejected. I recently ran into an interesting problem with two SIP phones trying to talk through Asterisk behind a (non-NAT) firewall. The problem was both phones were sending RTP to the Asterisk box but the firewall was blocking both RTP streams because Asterisk never sent any RTP out those ports. And the reason Asterisk hadn't sent RTP out those ports was because it was waiting for RTP from each of the two SIP phones. This was the classic chicken-and-egg scenario. I resolved it by opening up the firewall for the range of ports I had configured Asterisk to use for RTP. A better solution would be fore Asterisk to always send a starter RTP packet so that it can ensure that the firewall opens up. -brian Alexander Simeonidis wrote: Hello everybody, I'm new to Asterisk and I'm trying to configure the SIP side. I use Asterisk under the following configuration: SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution? Regards, Alex. Help STOP spam with the new MSN 8 http://g.msn.com/8HMAEN/2731??PS=47575 and get 2 months FREE*___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-doc] Conference hosting request for asterisk-doc
I'd be happy to host it, although it will be a toll call to Maryland. Contact me off list if you're interested. -brian Martin List-Petersen wrote: On Tue, 2004-05-11 at 21:52, Leif Madsen wrote: Afternoon all, Jared Smith and I would like to have a conference call Sunday evening to discuss the layout and direction of the Asterisk documentation project. We both feel that the layout we have is a good start, but it needs to be revised. I would consider what we have so far a first or second draft. What we need is someone to host the conference for us. We would like to have both VoIP access via SIP and IAX, and if possible, a 1-800 number. Why not using fwdnet for that ? They have several possibilites for dial-in in different countries and are accessible via iax and sip to everybody who subscribes (free). I can have a look into how we can arrange this and maybe put something together, that we also can use for future conferences. Martin List-Petersen martin at list-petersen dot net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musical interruptions
You know, I've seen something that may be related. We occasionally get DTMF inserted into the middle of a call, when no party on either end has pushed any buttons. I suspect something goes wrong and some data packet is mistakenly believed to contain out-of-band DMTF signalling and the * box is faithfully inserting it into the stream. But this is just a total guess. Perhaps you're seeing another manifistation of the same issue. -brian Mark Elkins wrote: Whilst on a call, I'm getting the following... -- Started music on hold, class 'default', on SIP/phone3-a7d5 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '#' in context 'default' -- Playing 'pbx-invalid' (language 'en') ie - without anyone pushing keys - I hear the music on Hold - as does the calling party. Are we somehow managing to sound like the tone for a '#' My BT100 phone is set up for DTMF=info This appears to happens quite randomly. Suggestions? I'm also getting quite a few... May 12 19:51:52 WARNING[98311]: chan_sip.c:542 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 152 (Non-critical Request) May 12 19:51:58 WARNING[98311]: chan_sip.c:542 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 153 (Non-critical Request) May 12 19:52:03 WARNING[4866068]: rtp.c:414 ast_rtp_read: RTP Read error: Resource temporarily unavailable May 12 19:52:03 WARNING[4866068]: rtp.c:414 ast_rtp_read: RTP Read error: Resource temporarily unavailable .. but am putting that down to running this extension (SIP Phone) over multiple 802.11 segments - in a semi-hostile environment. (I'm not the only person using 802.11 - there may be channel clashes) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line appearances
Hi Joseph, I'll assume you mean a 7960 with the SIP image... Yes, you can register the same SIP client multiple times. Each line appearance on a 7960 configured for SIP is a separate SIP client. Each can register with completely different SIP proxies (providers) or you can have several registrations for the same directory number (DN) so that instead of call waiting, additional calls appear at the next available line appearance. I can't answer your coded question since I always use g.711ulaw. -brian Joseph wrote: I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line appearances
That's exactly what it means. And, Asterisk will do the right thing. -brian Joseph wrote: Thanks for the help. Does that mean that say I had an extension 240 on line appearance one, I could make button 2,3,4 also register the same ext number? Would * care that there were multiple entries from the same ip for the same ext? Thanks again for the tips. On Tue, 2004-05-11 at 09:28, Brian Cuthie wrote: Hi Joseph, I'll assume you mean a 7960 with the SIP image... Yes, you can register the same SIP client multiple times. Each line appearance on a 7960 configured for SIP is a separate SIP client. Each can register with completely different SIP proxies (providers) or you can have several registrations for the same directory number (DN) so that instead of call waiting, additional calls appear at the next available line appearance. I can't answer your coded question since I always use g.711ulaw. -brian Joseph wrote: I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
It's not the switch. It's lightly loaded 100Mb. -brian Bisker, Scott (7805) wrote: What kind of switch do you have your phones plugged into? If your switch is highly loaded, or you are doing lots of multicast or broadcast, your SIP streams are going to suffer unless you are filtering that traffic at the port level or have separate VOIP VLANS. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Cuthie Sent: Friday, May 07, 2004 10:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway) It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
Ah, this reminds me that I forgot to mention that our network looks like this: Cisco --- SIP Asterisk IAX Aterisk IAX Asterisk PRI PSTN -brian Tom wrote: At 09:43 AM 5/7/2004, you wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). No dropout problems or choppy audio running Asterisk CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz P4 Supermicro server. Analog phones through our TDM400P do sound much better but the audio problems on our Cisco SIP phones are echo problems. People are working on solutions. Tom Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you close a VoicePulse Connect! account?
Anybody figured out how to close a VoicePulse Connect! account? As bad as their web site is at most other things, the notion of actually closing an account doesn't appear to have even been contemplated. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold spawns everlasting mpg123 processes
Actually, I think this is a feature. Rather than startup a new instance of mpg123 each time someone goes on hold, one instance per MOH class is created and used for all calls of that class. -brian Gavin Hamill wrote: Hullo :) I'm using CVS-04/23/04-23 from the stable 1.0 branch on kernel 2.6 - since I have no Digium h/w, I've just managed to get the zaprtc module to compile and run, so I thought the best way to test it would be via MoH. The MP3Player application works great .. exten = 6901,1,Answer exten = 6901,2,MP3Player(http://127.0.0.1:85/ES/28) This will play callers BBC Radio 4 from my local streaming setup, and when they hangup, the mpg123 process dies immediately. Perfect :) Unfortunately, the same cannot be said about: exten = 6900,1,Answer exten = 6900,2,MusicOnHold -- Accepting AUTHENTICATED call from 10.0.0.74, requested format = 2, actual format = 2 -- Executing Answer([EMAIL PROTECTED]/5, ) in new stack -- Executing MusicOnHold([EMAIL PROTECTED]/5, ) in new stack -- Started music on hold, class 'default', on [EMAIL PROTECTED]/5 Then I press 'Hangup' in IaxComm: -- Stopped music on hold on [EMAIL PROTECTED]/5 == Spawn extension (default, 6900, 2) exited non-zero on '[EMAIL PROTECTED]/5' -- Hungup '[EMAIL PROTECTED]/5' Alas, the 'mpg123' processes live on... 4049 pts/22 S 0:00 \_ asterisk -vc 4050 pts/22 S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 sample-hold.mp3 4063 pts/22 S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 sample-hold.mp3 In fact, if I redial ext 6900, I get played the MoH sample from the point at which mpg123 has reached in the mp3, rather than getting it from the start. The MusicOnHold docs say: Returns -1 on hangup. Never returns otherwise. I beg to differ :) Is this a bug, or have I made some fundamental mistake? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange IAX behaviors
I've been setting up a couple of * boxes with IAX trunking between
them. But I've been seeing some strange IAX behavior. Asterisk version
is latest CVS-04/21/04-18:10:19.
Here's what I'm doing: the boxes are peers, and I have setup my iax.conf
file to look something like this:
machine1
[iaxuser]
type=friend
username=iaxuser
secret=foo
auth=md5
context=iaxuser-incoming
host=machine2
machine2
[iaxuser]
type=friend
username=iaxuser
secret=foo
auth=md5
context=iaxuser-incoming
host=machine1
Now where things get weird is when I put the following line in the
extensions.conf file of machine1
switch = IAX2/iaxuser
authentication fails during dialing when machine2 sends the challenge
for authentication. The reason is that the username sent with the
challenge isn't the one defined for iaxuser.
Now, interestingly, if I change the switch line to include the user name
switch = IAX2/[EMAIL PROTECTED]
things work great!
Anybody have any idea what's happening here? Is this a bug? I would
think that the username would be implied by the definition for the
connection (as it seems to be for Dial(IAX2/iaxuser/${EXTEN})).
Another interesting thing that I've seen is that if a call comes in
through an IAX connection and then tries to do a remote diaplan
translation using 'switch', it seems to fail with a message stating that
there's no '[EMAIL PROTECTED]' (where is the number to match,
and 'incoming-context' is the context defined for calls from my IAX
provider). I've triple checked the dialplan, and I'm convinced it's not
the problem. Changing the incoming-context to use
Dial(IAX2/iaxuser/) works fine (notice that I don't need to
explicitly define the username here as I did with the 'switch' parameter).
Cheers,
Brian
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Re: [Asterisk-Users] Ok, Im confused
Not exactly :-)
Actually, you can get FWD to work through a NAT. SIP will send its
registration requests out the same port it expects a response on. This
will create a mapping in your NAT so that when INVITEs appear from FWD
the NAT can figure out which local host to send them to. Same thing
happens with SIP between the caller and the local machine. RTP does the
same trick too. I do this all the time with a Linksys router without any
difficulty.
Now things get a little more complicated if you have more than one SIP
device behind the firewall that's trying to talk to the outside world.
You'll need to configure each to use a unique SIP port, as well as a
unique range of RTP ports. If not, the NAT will see more than one
local_ip:proto:port tuple and will have to remap the port as the packet
leaves the router. But since the VIA header in the SIP packets will
refer to the original port, incoming SIP traffic will end up at the
wrong local host. With * you can avoid these problems by having all
traffic go through the * box. Do this by adding reinvite=no to your
sip.conf, and configure your SIP phones to use * as a proxy. Do not turn
on NAT in the SIP phones.
Actual firewalls (as apposed to just NATs like on a Linksys) may pose
additional problems depending on how they're configured. But that's
probably now what you're experiencing.
Cheers,
-brian
Scott Weis wrote:
The simple answer probably is, If you have a NAT firewall (like a linksys,
netgear, dlink, etc) it will not work.
If your linux machine is directly connected follow the instructions on the
wiki and it will work no problems. I could not get FWD to work at all until
I made my linux box the outside edge of my network.
Scott
- Original Message -
From: James H. Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 8:58 PM
Subject: Re: [Asterisk-Users] Ok, Im confused
You can post your .conf files.
But here is a guess at what you may need
replace FWD## with your freeworlddialup number and mypassword with
your freeworlddialup
password.
in sip.conf
context = from-fwd
register=FWD##:[EMAIL PROTECTED]/FWD##
[fwd]
type=friend
secret=mypassword
username=FWD##
host=fwd.pulver.com
in phone.conf
...
context=from-phone
...
in extensions.conf
[from-fwd]
exten = FWD##,1,Dial(Phone/phone0)
exten = i,2,Playback(invalid)
[from-phone]
exten = _.,1,SetCallerID(FWD##)
exten = _.,2,SetCIDName(FWD##)
exten = _.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _.,4,Playback(invalid)
exten = _.,5,Hangup
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: tmpm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 1:24 PM
Subject: Re: [Asterisk-Users] Ok, Im confused
Thanks Jim,
But that page started my trip off to confusionbeen theretried it
10
different ways...still no joy.
I'll go through it once again, maybe Im missing something, I dont know.
Im
about ready to boot the penguin to the curb...
I know its in there...I think Ive got it all configured, and I dial the
outbound strings, and get a fast busy...I know one stinking letter off,
and
its whacked...
HOW for example do I specify my one and only extension is the Internet
phone jack? Phone0?
Somehow theres got to be a tie-in...I cant find it.
been thru extensions.conf, phones.conf, sip.conf..etc.
groan..
At 18:40 4/21/2004, you wrote:
Look here:
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: tmpm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 11:50 AM
Subject: [Asterisk-Users] Ok, Im confused
Im totally a newbee at *
Im confused.
Ive got a FWD account, and it works on the winboxen. Ive got * up
and can
do the echotest etc, so its working.
I want to get FWD working, and all the pages ive seen on setup are
most
confusing.
Is FWD setup like IAXTEL? Do i plug in my FWD info in the same
places as
the IAXTEL stuff?
Ive been trying for a week now, and Im more lost than before.
Ive got a Internet phonejack card in the penguin, phone0, and all I
want to
do at this point is make and receive calls thru FWD using that
jackIll
plug the house in later...Ill learn the other stuff later. No
voicemail, no
BS, no dial thru least cost routing, or nightlines just make it
work as
a phone.
Im either more stupid than I think, or Im missing something major
here.
Ive got to the point the CLI shows me connected to FWD fine.(I
think)
Sip show users
Username Secret Authen Def. Context a/c
fwd.pulver.com secret md5,plaintext default no
Need some basic, stupidly simple scripts here...I dont need or want
to
dial
1-700 or *9 or any other crap, just make it work like the stupid
[Asterisk-Users] IAX config documentation
Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans - trunking - authentication - transfers But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options. There's the beginnings of a whitepaper on wiki, but it's self-contradictory in some places, largely incomplete, and just kind of ends abruptly. Yet, it mentions that growing contingent of IAX devices. Huh. Thanks -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX config documentation
I know that this stuff is. What I'm looking for is an overview of how these features work in the context of IAX. For instance, trunking is a concept I think we all get. But how do you use IAX to establish trunking between two switches? What's the effect of turning the transfer option on? How are dialplans shared between switches that are connected via IAX? What kinds of authentication are supported? How are keys managed? -brian Steven Critchfield wrote: On Mon, 2004-04-19 at 16:08, Brian Cuthie wrote: Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans This is extensions.conf work. Some of it can be shared via the switch command. - trunking Trunking is easy, think of it kind of like a channelized t1. It combines many calls into one packet with call data so as to reduce the overhead of each individual call having it's own resources. Specifically it cuts down on the overhead in IP, and allows you to reclaim some of the bandwidth for more calls. - authentication You do want to know who is trying to call you don't you? - transfers Allows you to get out of the middle of a call. My office loves these as our trunk lines are remote, and when we forward a call out to another trunk line, our local asterisk machine transfers the call back to the machine with trunk lines and removes the VoIP part of the loop. But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options. There is plenty of banter on the list and info scattered about that google will find for you than reading the source. Of course, you are free to bludgen yourself with the code if you so wish. There's the beginnings of a whitepaper on wiki, but it's self-contradictory in some places, largely incomplete, and just kind of ends abruptly. Yet, it mentions that growing contingent of IAX devices. Huh. Jump in and help finish it when you have read some and start to understand the missing parts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Fwd: Re: [Asterisk-Users] IAX config documentation]
Boy after really digging into this, I have discovered that there is more information about each of these topics than I previously realized. Strangely, searching the wiki on iax returns exactly nothing. But searching on iax2 does start to dig up some good stuff. Sorry for the hassle. Tough day. -brian Original Message Subject: Re: [Asterisk-Users] IAX config documentation Date: Mon, 19 Apr 2004 21:22:44 -0400 From: Brian Cuthie [EMAIL PROTECTED] To: [EMAIL PROTECTED] References: [EMAIL PROTECTED] [EMAIL PROTECTED] I know that this stuff is. What I'm looking for is an overview of how these features work in the context of IAX. For instance, trunking is a concept I think we all get. But how do you use IAX to establish trunking between two switches? What's the effect of turning the transfer option on? How are dialplans shared between switches that are connected via IAX? What kinds of authentication are supported? How are keys managed? -brian Steven Critchfield wrote: On Mon, 2004-04-19 at 16:08, Brian Cuthie wrote: Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans This is extensions.conf work. Some of it can be shared via the switch command. - trunking Trunking is easy, think of it kind of like a channelized t1. It combines many calls into one packet with call data so as to reduce the overhead of each individual call having it's own resources. Specifically it cuts down on the overhead in IP, and allows you to reclaim some of the bandwidth for more calls. - authentication You do want to know who is trying to call you don't you? - transfers Allows you to get out of the middle of a call. My office loves these as our trunk lines are remote, and when we forward a call out to another trunk line, our local asterisk machine transfers the call back to the machine with trunk lines and removes the VoIP part of the loop. But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options. There is plenty of banter on the list and info scattered about that google will find for you than reading the source. Of course, you are free to bludgen yourself with the code if you so wish. There's the beginnings of a whitepaper on wiki, but it's self-contradictory in some places, largely incomplete, and just kind of ends abruptly. Yet, it mentions that growing contingent of IAX devices. Huh. Jump in and help finish it when you have read some and start to understand the missing parts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote: When we receive or make a call to the outside they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote: I will try disallow=all, thanks, Nat is off. Sip.conf below. If I call from my mobile to the isdn number, the Cisco phone rings, I pick up, they can hear me, I can not hear them, if I transfer it to the Sipura I get AUDIO!! It is also happening over IAX with the Cisco phones. I followed a lot of the examples on loligo.com, which were a great help, but this is so hard to troubleshoot as I cannot see any errors in debug, asterisk thinks a good call is in progress. Anything internal is perfect. The CAPI works fine. Its just the audio from the other end. Every now and then I can hear a quick bit of sound. One in 20 calls may work. [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to allow=ulaw allow=alaw tos=lowdelay [20] type=friend username=20 secret=20 canreinvite=no host=dynamic mailbox=20 callerid=Cisco Phone 20 accountcode=20 qualify=yes context=sip Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 16 April 2004 18:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi - Asterisk à network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just to be clear, you need at least the following (or at least I did): sip.conf: nat=yes reinvite=no SIPDefault.conf (in your tftp directory) nat_enable=0 -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Spam
Tom Green wrote: Hi, Some people have suggested maintaining black lists and white lists to avoid spammers and allow legitimate callers into the network. However, the problem with this method is that the spammer's IP address might change due to DHCP. Today a spammer might get aaa.bbb.ccc.ddd and lets say that I put this address in my blacklist. To my annoyance, tomorrow a legitimate caller might get aaa.bbb.ccc.ddd and the spammer might get a different IP address. In the end, I end up blocking the legitimate caller also. Any ideas or thoughts to on this problem is appreciated. Thanks, Tom __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yeah, for a relatively modern protocol SIP has some surprisingly glaring omissions, such as: - certificate based authentication - encryption - NAT-awareness -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FAX?
Should FAX transmission generally work through Asterisk and a TDM400P connected through a PSTN gateway? At first blush I'd think that if they're all g.711uLaw encoded that it would work. But experience shows otherwise. Is there a better way to do FAX? -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan Format Strings
Try something like this:
exten = _9NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1${NPA}${EXTEN:1}
...
-brian
Nik Martin wrote:
In the absence of The Definitive Guide to Asterisk Dial Plans book, I'd
like to do something possibly unique with the formatting of extensions in my
dial plan, and am having trouble. We use VoicePulse connect, which gives us
local DID for inbound and outbound calls (even though DTMF tones are not
working in Voice Pulse Connect at the moment). To dial local numbers, you
have to dial the entire number, like 1 + area code + number. I'd like to
eliminate this by having the user just dial 9 + 7 digit number, and have
asterisk put the 1 + area code (which is in a variable in extensions.conf)
in front of it prior to sending the request to Voice Pulse. Is this
possible?
Thanks,
Nik Martin
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Re: [Asterisk-Users] Hunting S(n)IPs
Andrew Thompson wrote: [EMAIL PROTECTED] wrote: Another observation of something which doesn't work: exten = 3200,1,Dial(SIP/3200,20,tTr) exten = 3200,2,Playback(tt-weasels) exten = 3200,3,Hangup exten = 3200,102,Dial(SIP/3201,20,tTr) exten = 3200,103,Playback(tt-weasels) exten = 3200,104,Hangup exten = 3200,203,Dial(SIP/3202,20,tTr) exten = 3200,204,Playback(tt-weasels) exten = 3200,205,Hangup The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the first call has been answered. Therefore, Call#2 happily dials 3200 again, although 3200 is currently talking. I also tried to limit the number of calls going to the phone with outgoinglimit=1 in the sip.conf, but that makes no difference either. According to the wiki that functionality is broken. Two things: 1) Have you looked at call queue's? 2) I think you should have been looking at incominglimit, not outgoinglimit, or possibly both of them together in some combination. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I may be missing something here, but I'll make this suggestion just in case you haven't already considered it. Have your phone register multiple call appearances with the same DN. For instance, my 7960 has three appearances of 2205. Calls are automatically offered to the first available appearance, kind of like what you'd expect. I think this is the behavior you're looking for, but you may be trying to do it he hard way. Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
Tor Houghton wrote: On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote: Use IAX2, it is a better IAX protocol. Jeremy McNamara P.S. If you really must have it, dig thru the channels/Makefile, but there is zero reason to use it any longer. Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Probably a port collision on your NAT box. I believe that IAX and IAX2 use different ports. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePlus
Andrew Thompson wrote: Brian Cuthie wrote: I've been using VoicePlus for a few days now, and overall I'm fairly pleased. But one thing that truly scares me is that the drop-down box on their site where you re-charge your account has values that go all the way to $10,000.(!) I'm deathly afraid that one day in a drunken stupor I'll go to recharge for $30 and end up unwittingly adding $10K to my account. I've suggested that they remedy this, but it would be nice if other VP users who have similar feelings would contact them. Cheers, Brian Not to mention the transaction fees on a charge of that size would be about USD 200. That's really the place where you should have an open account, be able to do wire transfers, or fill in the blank your favorite overnight carrier them a certified check. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yeah, I agree. I did some digging and they look like they're just reselling Transbeam service. I'm going to continue to look for a new service provider. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT appologies to list
There's more than a little irony here, given that one of their products is called Email Blaster. -brian Linus Surguy wrote: [I'm sorry to trouble the list with this, but this is the only way I know to contact the person concerned] This message is for Stephen Karrington - it appears that you have over-agressive 'spam' filters and we can no longer email you. Please rectify this if we are to have meaningful conversation! The original message was received from Linus Surguy [EMAIL PROTECTED] - This message has been blocked by our spam filter. - - If this has been a mistake, please contact- - the recipient through other means.- Undelivered message for: Stephen Karrington [EMAIL PROTECTED] (reason: 550 Spam blocked) - Transcript of session follows - 550 Spam blocked DATA From [EMAIL PROTECTED] Mon Apr 12 15:23:04 2004 Received: from idoru.world.co.uk (idoru.world.co.uk [213.166.5.65]) by sky.dreamtime.net (8.12.10/8.12.10) with ESMTP id i3CJN3Ve093074 for [EMAIL PROTECTED]; Mon, 12 Apr 2004 15:23:04 -0400 (EDT) Received: from news.world.co.uk (office-gw.magrathea-systems.co.uk [213.162.109.118]) by idoru.world.co.uk (8.10.2/8.10.2) with ESMTP id i3CJMwc15379 for [EMAIL PROTECTED]; Mon, 12 Apr 2004 20:22:58 +0100 Received: from molly (molly.world.co.uk [10.0.0.2]) by news.world.co.uk (8.12.0/8.12.0) with SMTP id i3CJMtDk006598 for [EMAIL PROTECTED]; Mon, 12 Apr 2004 20:22:55 +0100 Message-ID: [EMAIL PROTECTED] From: Linus Surguy [EMAIL PROTECTED] To: Stephen Karrington [EMAIL PROTECTED] References: [EMAIL PROTECTED] [EMAIL PROTECTED] Subject: Re: Re[4]: [Asterisk-Users] Who has access numbers in the UK and Germany? Date: Mon, 12 Apr 2004 20:22:59 +0100 MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4522.1200 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Question
And buy a bigger disk while you're at it :-) They're under a $100 and then you can let your users create their own outgoing messages. -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 11, 2004 12:20 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicemail Question On 11 Apr 2004 at 18:16, Paul Tyreman wrote: What does that do then ? snipped du -sh /var/spool/asterisk/vm/* At the command line, do man du You will have to know a bit about the operating system, this is not point and click. John Chapman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePlus
Title: VoicePlus I've been using VoicePlus for a few days now, and overall I'm fairly pleased. But one thing that truly scares me is that the drop-down box on their site where you re-charge your account has values that go all the way to $10,000.(!) I'm deathly afraid that one day in a drunken stupor I'll go to recharge for $30 and end up unwittingly adding $10K to my account. I've suggested that they remedy this, but it would be nice if other VP users who have similar feelings would contact them. Cheers, Brian
RE: [Asterisk-Users] No ringing tone with IAXY (and other bits and bobs)
What version of the Asterisk code are you running? 1_0 stable is definitely
broken wrt ringback, and the latest stuff seems really broken in all kinds
of ways. After seeing that others were having similar problems, and that
someone had solved many of them by rolling back to the CVS version from 3/5,
I tried the same and things are working marvelously (well, mostly).
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Orme
Sent: Saturday, April 10, 2004 6:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No ringing tone with IAXY (and
other bits and bobs)
Hi!
I'm really hope you can help me solve a little mystery, the
mystery is probably just my misunderstanding ! sorry...
I've got an iaxy talking to my * box which connects to two providers.
I'm running the stable release of the pbx.
The only thing is that when dialling from the iaxy the
ringing tone isn't heard while calling someone - you just
hear silence then, they either answer or they don't on the remote end.
From my extensions.conf is the following - I tried putting the ,r in
and
it doesn't help. Is there some other option I could try here ?
Also I'm getting quite a bit of echo noticed at the remote
end as well as the iaxy end. All lines are digital, I guess
only the jitter buffer is there to be tweaked to try and help ?
There is also this echo problem with the sipura, but not with
an ATA186 or snom. The lack of a ringing tone is only with the iaxy.
The Answer,Hangup lines were to solve 'busy' situations with
SIP phones, without this or even with 'Congestion' they just
rang forever if a number was busy. They seem to need the
'Answer' line.
If you know a nicer or more correct way for me to do this
please let me know as most times the SIP phone user will hear
half a ring and then the hangup noise generated by the SIP
device when a number they call is busy.
Many thanks!!
Chris
PS please Cc: me a copy as well as to the list in case I miss
it - Thanks.
extensions.conf
exten = _00.,1,AbsoluteTimeout(3600)
exten = _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
exten = _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
exten = _00.,104,Answer
exten = _00.,105,Hangup
iax.conf
[iaxy]
type=friend
accountcode=iaxy
disallow=all
;;allow=adpcm
allow=ulaw
username=iaxy
secret=xxx
auth=md5
nat=yes - nat=1 ??
notransfer=yes -this doesn't seem to work, perhaps in the
wrong order?
host=dynamic
qualify=1
Is the definitive order these should be in listed anywhere as
I know it really seems critical and lines can be ignored if
they're not in spot on the right order?
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RE: [Asterisk-Users] No ringing tone with IAXY (and other bits and bobs)
Sure. I used this to get the 3/5 version:
cvs co -D 20040305 zaptel asterisk
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Saturday, April 10, 2004 9:13 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No ringing tone with IAXY (and
other bits and bobs)
Brian,
I need to roll back to an earlier version to identify a
different problem, but I dont have the cvs checkout command
string that includes a date. Can you post how to do that please?
Rich
What version of the Asterisk code are you running? 1_0 stable is
definitely broken wrt ringback, and the latest stuff seems really
broken in all kinds of ways. After seeing that others were having
similar problems, and that someone had solved many of them
by rolling
back to the CVS version from 3/5, I tried the same and
things are working marvelously (well, mostly).
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Orme
Sent: Saturday, April 10, 2004 6:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No ringing tone with IAXY (and
other bits
and bobs)
Hi!
I'm really hope you can help me solve a little mystery,
the mystery
is probably just my misunderstanding ! sorry...
I've got an iaxy talking to my * box which connects to
two providers.
I'm running the stable release of the pbx.
The only thing is that when dialling from the iaxy the
ringing tone
isn't heard while calling someone - you just hear silence
then, they
either answer or they don't on the remote end.
From my extensions.conf is the following - I tried
putting the ,r
in and
it doesn't help. Is there some other option I could try here ?
Also I'm getting quite a bit of echo noticed at the remote end as
well as the iaxy end. All lines are digital, I guess only the
jitter buffer is there to be tweaked to try and help ?
There is also this echo problem with the sipura, but not with an
ATA186 or snom. The lack of a ringing tone is only with the iaxy.
The Answer,Hangup lines were to solve 'busy' situations with SIP
phones, without this or even with 'Congestion' they just rang
forever if a number was busy. They seem to need the
'Answer' line.
If you know a nicer or more correct way for me to do this
please let
me know as most times the SIP phone user will hear half a
ring and
then the hangup noise generated by the SIP device when a
number they
call is busy.
Many thanks!!
Chris
PS please Cc: me a copy as well as to the list in case I
miss it -
Thanks.
extensions.conf
exten = _00.,1,AbsoluteTimeout(3600) exten =
_00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
exten = _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
exten = _00.,104,Answer
exten = _00.,105,Hangup
iax.conf
[iaxy]
type=friend
accountcode=iaxy
disallow=all
;;allow=adpcm
allow=ulaw
username=iaxy
secret=xxx
auth=md5
nat=yes - nat=1 ??
notransfer=yes -this doesn't seem to work, perhaps in the wrong
order?
host=dynamic
qualify=1
Is the definitive order these should be in listed
anywhere as I know
it really seems critical and lines can be ignored if
they're not in
spot on the right order?
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RE: [Asterisk-Users] PC based Switchboard application
Probably for the same reason you charge for your services. Software takes time and skill to write. And while I'm grateful that people like Mark release their apps to us as open source or under GPL, I don't begrudge anyone from wanting to actually make a living. -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Saturday, April 10, 2004 9:40 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PC based Switchboard application Pertti Pikkarainen [EMAIL PROTECTED] wrote: We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. Why not? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1.0_stable is or isn't? (Was: No ringing tone...)
When I installed 1_0_STABLE, ringback stopped working completely on all calls through the TDM400P. I can't recall if the SIP phones stopped generating ringback also. Latest builds (as of yesterday) seem to have problems with dropouts, especially with IAX connections. I was seeing dropouts and repeated packets (think Max Headroom) over IAX channels. Checking voicemail from a SIP phone resulted in dropouts pretty consistently when it was playing menus. Now, mind you, I'm not really complaining, since this is not released code. This is from the development CVS tree. But, in my experience it does seem to be broken. However, 3/5 seems to work well for me. Although I am having some trouble with Zapateller. Cheers, brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Klepfer Sent: Saturday, April 10, 2004 11:52 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 1.0_stable is or isn't? (Was: No ringing tone...) Brian Cuthie wrote: What version of the Asterisk code are you running? 1_0 stable is definitely broken wrt ringback, and the latest stuff seems really broken in all kinds of ways. After seeing that others were having similar problems, and that someone had solved many of them by rolling back to the CVS version from 3/5, I tried the same and things are working marvelously (well, mostly). I've been swamped at work and heven't been able to keep up with the version discussions or monitor asterisk-cvs closely. Could you qualify your statement above about 1-0_stable being broken? I'm running 1.0 stable (CVS-03/20/04-22:33:52) here at work and have noticed faxing over SIP much more stable, but a couple of momentary dropouts on outside calls (GS bt101 - x100p POTS), usually after silence in the conversation. (I *have* noticed RAM almost completely filled, but no swap used...a reboot freed a bunch and I think that fixed some issues. We're a small company and restarting * or rebooting the server isn't that big a deal.) Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with Zpateller on incoming external calls
Title: Problems with Zpateller on incoming external calls I've setup the following in extensions.con: exten = 2200,1,Ringing exten = 2200,2,Wait(2) exten = 2200,3,Answer exten = 2200,4,Zapateller exten = 2200,5,Macro(stdexten,2205,SIP/2205) This works as expected if I dial from a SIP phone on my desk. However, if I dial in from the PSTN (through a SIP provider) it fails while trying to play ths SIT with: Apr 8 18:53:12 WARNING[1209269552]: rtp.c:407 ast_rtp_read: RTP Read error: Resource temporarily unavailable Any idea what's going on? My suspicion is that the PSTN gateway hasn't setup an audio path yet, although I thought Answer would do that. Cheers, Brian
RE: [Asterisk-Users] Problems with Zpateller on incoming external calls
Tried that, and no go. There's something wrong with Zapteller. It works fine on internal calls, but the only way I can get it to work on external calls (through a SIP/PSTN gateway, no Zap hw necessary) is to first play a message. For instance, this works: exten = 2200,1,Playback(ss-noservice) exten = 2200,2,Zapateller exten = 2200,3,Dial(SIP/2205) -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Friday, April 09, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with Zpateller on incoming external calls Brian Cuthie wrote: I've setup the following in extensions.con: exten = 2200,1,Ringing exten = 2200,2,Wait(2) exten = 2200,3,Answer exten = 2200,4,Zapateller exten = 2200,5,Macro(stdexten,2205,SIP/2205) This works as expected if I dial from a SIP phone on my desk. However, if I dial in from the PSTN (through a SIP provider) it fails while trying to play ths SIT with: Apr 8 18:53:12 WARNING[1209269552]: rtp.c:407 ast_rtp_read: RTP Read error: Resource temporarily unavailable Any idea what's going on? My suspicion is that the PSTN gateway hasn't setup an audio path yet, although I thought Answer would do that. Cheers, Brian I don't have a zap device to test on, but can you do Ringing before you Answer? - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
Can this Frtiz card be used in the US? -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jakob Strebel Sent: Thursday, April 08, 2004 3:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI Jean-Marie, Hi, First, here is my config: Kernel version 2.4.25 on a Fedora distro, Asterisk and a Fritz! Isdn PCI Card (v2). I try to make the CAPI drivers deal with Asterisk but I can't try to figure out to get of this issue. As I see, Fritz modules are integrated with the kernel, so I directly loaded the 'hisax_fcpcipnp' module from it. I install also Capi modules by downloading archives of the web (make config - make install - insmod...). It is my understanding not to use hisax. Use chan_capi instead. http://www.voip-info.org/tiki-index.php?page=Asterisk+How+to+c onnect+with+CAPI http://www.junghanns.net/asterisk/page1.html I have this working. jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TigerJet ISDN card
Title: TigerJet ISDN card Is there any Linux/* support for the TigerJet ISDN card? -brian
[Asterisk-Users] IAXTel toll-free gateway
Title: IAXTel toll-free gateway Is anyone else having trouble placing toll-free calls though IAXTel lately? Mine just stopped working yesterday, yet I seem to be able to make 1-700 calls. -brian 1-700-676-3830
RE: [Asterisk-Users] ISDN BRI solution for USA
I'm also looking for the same thing: ISDN-BRI U interface. Thanks. -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alfred R. Nurnberger Sent: Wednesday, April 07, 2004 9:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ISDN BRI solution for USA I am looking for a ISDN BRI card (u-INTERFACE) to connect * to a US 5ESS switch (Qwest). According to Qwest they support CNAME delivery on their 5ESS switches. Does * chan_capi support CNAME ? Regards. Alfred. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP -- PSTN gateways
Title: SIP -- PSTN gateways So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if I'm already on a call and another comes in will they attempt to deliver it? Thanks -brian
RE: [Asterisk-Users] mpg123 issue and solution
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Maresca It might be a good idea to move away from mpg123 as it is no longer supported and there are bound to be more problems like this. MAD seems to be what everyone is migrating to... At the very least, not hardcoding a player into the codebase would probably be a good idea (if it is hardcoded, I couldn't find a config file for it...). The app is hard-coded. Take a look at res_musiconhold.c (in the res sub directory of the Asterisk source). At the vere least you can change the source and recompile if you want to use a different player app. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Disambiguating incoming IAXTel calls
Hmmm... So I tried this with an iax.conf file that looks like this: register = username1:[EMAIL PROTECTED] register = username2:[EMAIL PROTECTED] ; calls coming into 1-700-xxx-xxx1 [username1] type=user context=iaxtel-incoming-username1 auth=rsa inkeys=iaxtel ; calls coming into 1-700-xxx-xxx2 [usrname2] type=user context=iaxtel-incoming-username2 auth=rsa inkeys=iaxtel [iaxtel] type=user context=iaxtel-incoming auth=rsa inkeys=iaxtel The problem is that all calls are coming in as user [iaxtel]. I've verified this by turning iax2 debug on and looking at the traces. Am I doing somethig wrong? Or can you just not get there from here? Cheers, -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Cross Sent: Tuesday, April 06, 2004 2:08 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Disambiguating incoming IAXTel calls On Mon, 5 Apr 2004, Brian Cuthie wrote: I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. You should have two type=user entries in your iax.conf, one for each account. Make sure each one specifies a different context, and set up extensions.conf appropriately. Note that I have not done this myself, and there was mailing list discussion on this topic in the last couple of months. Hit the archives for more information. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Brian Cuthie wrote: Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that don't emanate from a SIP phone here at my location. Asterisk SIP does not support silence suppression. In fact, using Silence suppression on an inbound RTP stream will lead to problems, since Asterisk takes timing from inbound RTP streams. Yeah, funny thing is I saw this problem just last night while messing around with music on hold. I had VAD on the SIP phone on and the MOH would stop unless I talked. I thought it was quite weird when it happened; now it makes sense. I've heard that Asterisk derives its timing in strange ways, but I've been wondering why it doesn't use the machine's clock (real-time interrupt, not wall-clock). -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Buzzing on TDM400P FXS?
Haven't seen this, but I do hear a loud click about 5 seconds into any call involving a TDM400P port. Seems like something might not be quite right with the Zap driver. -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Monday, April 05, 2004 1:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Buzzing on TDM400P FXS? I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for buzzing on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and IAX to NuFone, so I'm pretty sure that it's happening on the FXS end. Here's that chunk of zapata.conf: context=inside-analog signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes cancallforward=yes callreturn=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=0 immediate=no musiconhold=yes usecallerid=yes callerid=Analog Phone 2201 mailbox=2201 channel = 2 Does anyone have any suggestions on where to start looking? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disambiguating incoming IAXTel calls
Title: Disambiguating incoming IAXTel calls I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. Thanks -brian
RE: [Asterisk-Users] Auto connect to voicemail
I use something like this:
exten = 8500,1,Ringing
exten = 8500,2,Wait,1
exten = 8500,3,VoicemailMain(s${CALLERIDNUM})
Basically, this rings the phone for once second (thus setting up the audio
path), then goes to voicemail without requiring the password. Leave out the
's' to have VM prompt for the password.
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian Rathman
Sent: Monday, April 05, 2004 3:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Auto connect to voicemail
I have the voicemail setup working in that I get the MWI and
it emails the message correctly. When I pressed the MWI
button on my SNOM 200, it dials into the voicemail system and
prompts me for a mailbox and password. I know there is a way
to automatically connect directly into the mailbox via the
extension.conf file, but I can not find the documentation I
am looking for in reference to variables and macros for the
extensions file. Can someone please help me with this issue?
Thanks,
Brian
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RE: [Asterisk-Users] Stable Relase Broken ?
I ran into the same problem. It seems to be fixed in later builds. -brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, April 05, 2004 5:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Stable Relase Broken ? All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?
Title: Silence suppression on SIP calls generated from Asterisk? Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that don't emanate from a SIP phone here at my location. Thanks -brian
[Asterisk-Users] No ringback
Title: No ringback I just configured Asterisk on a new machine, and other things seem to be working fine, I don't get any audible ringback when dialing calls from a SIP phone or a standard phone connected through a TDM400P. What am I doing wrong here? Thanks -brian
RE: [Asterisk-Users] No ringback
Title: No ringback Thanks. Actually,I got the latest from the cvs repository and it's fixed there, too. I suspect that it got broken at some point briefly before someone fixed it. -brian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gene KochanowskySent: Saturday, April 03, 2004 5:25 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] No ringback I had a similar problem. What I did what checked out the version before 03-02-2004. Some change after that date is causing the problem. Gene From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian CuthieSent: Saturday, April 03, 2004 4:32 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] No ringback I just configured Asterisk on a new machine, and other things seem to be working fine, I don't get any audible ringback when dialing calls from a SIP phone or a standard phone connected through a TDM400P. What am I doing wrong here? Thanks -brian
[Asterisk-Users] ISDN BRI-U card suggestion for use in USA
Title: ISDN BRI-U card suggestion for use in USA Hello, I'm looking for an ISDN BRI-U interface for use in the US. I'm primarily interested in using the BRI as a trunking interface into the PSTN with Asterisk. Naturally, cheaper is better. I currently use a Nortel Norstar system with BRI-U trunks, and really like the digital PSTN interface. Would really like to replace the whole mess with Asterisk but want to keep the ISDN trunks. Since this for use in a small SOHO installation, PRI is kind of out of the question. Any suggestions?? Thanks -brian
[Asterisk-Users] DTMF not being detected on PhoneJack-lite
Title: DTMF not being detected on PhoneJack-lite I'm trying to get a PhoneJack-Lite to work on my Asterisk box. I've actually gdb'd the code and it looks like I'm never getting any DTMF events. Does the PhoneJack-Lite work with Asterisk? Are there some limitations with using it that I may be bumping up against? Thanks -brian
RE: [Asterisk-Users] Cisco 7960 SIP Images
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Lawrence Sent: Tuesday, March 30, 2004 12:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images ... I have no problem with the idea of paying cisco for software that they write. In fact I have no problem with with paying for software full stop. But I'd love to have enough money to sue them if that software proved to have security issues or proved to be not fit for purpose - eg if a phone had a bug in its implementation of SIP. If people/companies want to charge for software fine (after all it takes time/money to develop) but they should be willing to take the responsibility that goes with it. Most companies don't - at least if you cantact cisco with a problem then they'll do their best to fix it or at least come up with a work-around, which is more than a certain other companies do. Jon I don't have a problem paying for updates, even if they include bug fixes. I write software for a living, and it's an imperfect art. My beef with Cisco is that the software license doesn't travel with the device. Without the license you can't buy an upgrade even if you want to. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_phone problems
Title: Chan_phone problems Hi All, I'm really new to Asterisk, and I'm having a little trouble with my test setup. Things are pretty simple so far: Linux 2.4 kernel (Redhat 9) Linux PhoneJack-Lite interface What happens is that I get dialtone, but dialing doesn't seem to do anything. I neither get connected to whatever I dialied, nor do I lose dialtone. I'm pretty sure I'm doing something dumb, and any help is greatly appreciated. One general question I have regards interfaces and incoming calls. I think I basically understand the dialplan concept, but it seems to deal soley with where a call goes. What I can't seem to figure out is how Asterisk configures input channels. Cheers, Brian
RE: [Asterisk-Users] Cisco 7960 SIP Images
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roderick Montgomery Sent: Monday, March 29, 2004 4:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images ... ### ### Hardware != Software ### Cisco IOS Software, phone firmware, etc. is normally bundled with hardware at the time of purchase, because, frankly, the hardware isn't really of much use without software. You may resell the hardware (which, looking at eBay, happens frequently), but the software license DOES NOT transfer from one end user to another. There are only a few exceptions to this rule, such as for business affiliates, mergers, acquisitions, lease buyouts, and outsourcing arrangements. Frankly, this is a horrible policy. It's designed to eliminate the market for used gear so that vendors can force people to buy new equipment. Frankly, anyone with this business model should be ashamed. And anyone buying equipment under such circumstances should beware. The assets they think they're purchasing today have substantially less value than they think since they can't effectively resell them when they're no longer needed. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP Images
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, March 29, 2004 7:11 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP Images It's just the rule of the game, and the game plan is called by the author (not the user). Its not a lot different then 80% of the software vendors charging a large fee to upgrade when the first digit changes (eg, v1.x to v2.x), just different words. No, it's hugely different. We're not talking about support and ongoing maintenance releases, we're talking about the right to use the software already in the used box you jusy bought. It's just wrong, and the only thing that keeps them from doing it with the hardware is that the FTC would come after them for restraint of trade. Since SW is considered IP and is 'licsensed' rather than sold, all the normal rules don't apply. What I suspect large customers do is negotiate contracts that include a transferable software license. As always it's the little guys who get screwed. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
